[asterisk-commits] file: branch file/pimp_sip_media r381155 - /team/file/pimp_sip_media/channels/

[SVN commits to the Asterisk project]

Author: file
Date: Sun Feb 10 15:20:58 2013
New Revision: 381155

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=381155
Log:
Fix a bug where the RTP instance returning no frame would cause a crash.

Modified:
    team/file/pimp_sip_media/channels/chan_gulp.c

Modified: team/file/pimp_sip_media/channels/chan_gulp.c
URL: http://svnview.digium.com/svn/asterisk/team/file/pimp_sip_media/channels/chan_gulp.c?view=diff&rev=381155&r1=381154&r2=381155
==============================================================================
--- team/file/pimp_sip_media/channels/chan_gulp.c (original)
+++ team/file/pimp_sip_media/channels/chan_gulp.c Sun Feb 10 15:20:58 2013
@@ -229,7 +229,7 @@
 		f = &ast_null_frame;
 	}
 
-	if (f->frametype == AST_FRAME_VOICE) {
+	if (f && f->frametype == AST_FRAME_VOICE) {
 		if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &f->subclass.format))) {
 			ast_debug(1, "Oooh, format changed to %s\n", ast_getformatname(&f->subclass.format));
 			ast_format_cap_set(ast_channel_nativeformats(ast), &f->subclass.format);